Conventionally, apparatuses and methods for transmitting encoded voice via a packet-switching network are known, including, for example, a “Packet-Switching Scheme” (refer to Patent Document 1) and a “Packet Communication Method” (refer to Patent Document 2).
Communication apparatuses to be used in a packet-switching network include those compliant with ATM (Asynchronous Transfer Mode) and IEEE802.3. The recent popularization of high-speed Internet access via cable television, ADSL (Asymmetric Digital Subscriber Line), optical fiber, and the like has led to the practical realization of VoIP (Voice over IP) apparatuses that transmit/receive voice data and therefore enable verbal communication over IP (Internet Protocol). Such apparatuses are also referred to as IP telephones.
The receiver of such a voice transmission apparatus is provided with a jitter absorbing buffer for absorbing transmission delay jitters that occur in the packet-switching network.
FIG. 13 is a block diagram illustrating a conventional packet-switching scheme. As shown in FIG. 13, in a conventional packet-switching scheme, a delay-difference absorbing buffer is provided at the receiver, the transmitter adds sequence numbers to packets belonging to the same call, and the receiver stores received packets in the delay-difference absorbing buffer while monitoring the sequence numbers of the received packets. In addition, reading of received packets from the delay-difference absorbing buffer starts at a constant rate upon the lapse of a predetermined period of time from the moment the first packet belonging to the relevant call is received.
When there is a loss in the sequence numbers of received packets, the receiver stores the same number of excess packets as the lost sequence numbers into the delay-difference absorbing buffer. In addition, when an underflow of the buffer occurs, the excess packets are inserted and replayed, and when packets having sequence numbers which should have been replayed are received during an underflow after insertion, such packets are discarded. Furthermore, when the delay-difference absorbing buffer overflows, the overflowed packets are discarded.
Accordingly, it is now possible to read and replay packets from the buffer at a certain timing without varying the playback timing. An excess packet is also called a dummy packet and is a packet that is either silent, contains background noise, or a repetition of a previously received packet. However, when the delay-difference absorbing buffer overflows due to a premature arrival of packets, the overflowed packets are discarded and the playback timing is altered. When a loss in sequence numbers occurs due to packet discarding and the like, excess packets are stored in the buffer.
In FIG. 13, when packets are received from a receiving line 1, a sequence number checking circuit 2 checks the sequence numbers of received packets. When there is a skip in the sequence numbers, the sequence number checking circuit 2 notifies a dummy packet generating circuit 3 by just that much, and in the case of voice, dummy packets such as silent packets, background noise packets and the like are to be written into a delay-difference absorbing buffer memory 4.
In addition, when the delay-difference absorbing buffer memory 4 is full and is unable to store received packets, the sequence number checking circuit 2 discards such packets. Furthermore, the sequence number checking circuit 2 detects a packet to arrive first after call setup is performed and notifies the same to a timer 5, and after a stochastically sufficiently long time D, starts consecutive playback of packets from the delay-difference absorbing buffer memory 4.
On the other hand, when the delay-difference absorbing buffer memory 4 becomes empty and an underflow occurs, dummy packets from the dummy packet generating circuit 3 are replayed and the occurrence of the underflow is notified to the sequence number checking circuit 2. When the relevant packets are subsequently received, the sequence number checking circuit 2 immediately discards such packets. The value of the time D, call setup information, and the like are to be notified to the respective circuits from a control circuit 6.
Currently, amidst the popularization of wireless LANs and mobile phone networks, demands are being made for realizing VoIP on such a wireless communication channel. With a wireless communication channel, communication quality may temporarily decline and, as a result, communication may be lost for a relatively long period of time. While the behavior at this time depends on the wireless communication protocol of the wireless communication channel, descriptions thereof are given in, for example, “Data Communication Apparatus” (refer to Japanese Patent Laid-Open No. 2006-101339) and “Data Communication Apparatus” (refer to Japanese Patent Laid-Open No. 2006-101340) by the present inventor.
That is, when a large number of packets awaiting transmission are transmitted after a relatively long communication breakdown, in some cases, the transmission causes downstream congestion or strains the processing capability of the receiver. This phenomenon is called a delay spike.
In addition, conventionally, in response to increasing demands for voice packet transmission, wireless communication apparatuses provided with QoS (Quality of Service) control in which control is executed in correspondence to service class have been emerging. With such apparatuses, since delay is reduced while errors and discarding are allowed in regards to voice packets, occurrences of delay spikes can be prevented.    Patent Document 1: Japanese Patent Laid-Open No. 01-029141    Patent Document 2: Japanese Patent Laid-Open No. 04-100454